2015-03-30 19:00:44 +02:00
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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2015-11-26 04:44:54 -08:00
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#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
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2015-03-30 19:00:44 +02:00
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#include "webrtc/base/checks.h"
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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#include "webrtc/base/format_macros.h"
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2015-11-26 04:44:54 -08:00
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#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
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2015-10-28 18:17:40 +01:00
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#include "webrtc/system_wrappers/include/trace.h"
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2016-10-07 07:07:28 +02:00
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#include "webrtc/voice_engine_configurations.h"
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2015-03-30 19:00:44 +02:00
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namespace webrtc {
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namespace acm2 {
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namespace {
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// Check if the given codec is a valid to be registered as send codec.
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2015-11-18 08:27:51 -08:00
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int IsValidSendCodec(const CodecInst& send_codec) {
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2015-03-30 19:00:44 +02:00
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int dummy_id = 0;
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if ((send_codec.channels != 1) && (send_codec.channels != 2)) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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"Wrong number of channels (%" PRIuS ", only mono and stereo "
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"are supported)",
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2015-11-18 08:27:51 -08:00
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send_codec.channels);
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2015-03-30 19:00:44 +02:00
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return -1;
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}
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2015-11-06 14:28:00 -08:00
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auto maybe_codec_id = RentACodec::CodecIdByInst(send_codec);
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if (!maybe_codec_id) {
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"Invalid codec setting for the send codec.");
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return -1;
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}
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// Telephone-event cannot be a send codec.
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if (!STR_CASE_CMP(send_codec.plname, "telephone-event")) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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"telephone-event cannot be a send codec");
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return -1;
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}
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2015-11-06 14:28:00 -08:00
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if (!RentACodec::IsSupportedNumChannels(*maybe_codec_id, send_codec.channels)
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.value_or(false)) {
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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"%" PRIuS " number of channels not supportedn for %s.",
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2015-03-30 19:00:44 +02:00
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send_codec.channels, send_codec.plname);
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return -1;
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}
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2015-11-06 14:28:00 -08:00
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return RentACodec::CodecIndexFromId(*maybe_codec_id).value_or(-1);
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2015-03-30 19:00:44 +02:00
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}
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2015-05-07 15:49:23 +02:00
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bool IsOpus(const CodecInst& codec) {
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return
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#ifdef WEBRTC_CODEC_OPUS
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!STR_CASE_CMP(codec.plname, "opus") ||
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#endif
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false;
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}
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2015-03-30 19:00:44 +02:00
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} // namespace
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2015-12-16 04:19:08 -08:00
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CodecManager::CodecManager() {
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2015-03-30 19:00:44 +02:00
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thread_checker_.DetachFromThread();
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}
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2015-04-13 09:31:16 +02:00
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CodecManager::~CodecManager() = default;
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2015-03-30 19:00:44 +02:00
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2015-12-16 04:19:08 -08:00
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bool CodecManager::RegisterEncoder(const CodecInst& send_codec) {
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2015-09-17 00:24:34 -07:00
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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2015-11-18 08:27:51 -08:00
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int codec_id = IsValidSendCodec(send_codec);
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2015-03-30 19:00:44 +02:00
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// Check for reported errors from function IsValidSendCodec().
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if (codec_id < 0) {
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2015-12-16 04:19:08 -08:00
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return false;
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2015-03-30 19:00:44 +02:00
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}
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int dummy_id = 0;
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2015-11-23 04:30:52 -08:00
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switch (RentACodec::RegisterRedPayloadType(
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&codec_stack_params_.red_payload_types, send_codec)) {
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kOk:
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2015-12-16 04:19:08 -08:00
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return true;
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kBadFreq:
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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2015-11-18 07:32:49 -08:00
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"RegisterSendCodec() failed, invalid frequency for RED"
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" registration");
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2015-12-16 04:19:08 -08:00
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return false;
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kSkip:
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break;
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}
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2015-11-23 04:30:52 -08:00
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switch (RentACodec::RegisterCngPayloadType(
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&codec_stack_params_.cng_payload_types, send_codec)) {
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kOk:
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2015-12-16 04:19:08 -08:00
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return true;
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kBadFreq:
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, dummy_id,
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2015-11-18 07:32:49 -08:00
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"RegisterSendCodec() failed, invalid frequency for CNG"
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" registration");
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2015-12-16 04:19:08 -08:00
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return false;
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2015-11-18 07:32:49 -08:00
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case RentACodec::RegistrationResult::kSkip:
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break;
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2015-03-30 19:00:44 +02:00
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}
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2015-12-16 04:19:08 -08:00
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if (IsOpus(send_codec)) {
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2015-12-01 04:45:04 -08:00
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// VAD/DTX not supported.
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codec_stack_params_.use_cng = false;
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2015-05-07 15:49:23 +02:00
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}
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2015-03-30 19:00:44 +02:00
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2015-12-16 04:19:08 -08:00
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send_codec_inst_ = rtc::Optional<CodecInst>(send_codec);
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2016-06-23 03:58:36 -07:00
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recreate_encoder_ = true; // Caller must recreate it.
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2015-12-16 04:19:08 -08:00
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return true;
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2015-03-30 19:00:44 +02:00
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}
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2015-12-16 04:19:08 -08:00
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CodecInst CodecManager::ForgeCodecInst(
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const AudioEncoder* external_speech_encoder) {
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CodecInst ci;
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ci.channels = external_speech_encoder->NumChannels();
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ci.plfreq = external_speech_encoder->SampleRateHz();
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ci.pacsize = rtc::CheckedDivExact(
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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static_cast<int>(external_speech_encoder->Max10MsFramesInAPacket() *
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2015-12-16 04:19:08 -08:00
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ci.plfreq),
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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100);
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2015-12-16 04:19:08 -08:00
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ci.pltype = -1; // Not valid.
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ci.rate = -1; // Not valid.
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2015-05-18 14:52:29 +02:00
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static const char kName[] = "external";
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2015-12-16 04:19:08 -08:00
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memcpy(ci.plname, kName, sizeof(kName));
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return ci;
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2015-03-30 19:00:44 +02:00
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}
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bool CodecManager::SetCopyRed(bool enable) {
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2015-11-23 04:30:52 -08:00
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if (enable && codec_stack_params_.use_codec_fec) {
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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"Codec internal FEC and RED cannot be co-enabled.");
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return false;
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}
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2015-12-16 04:19:08 -08:00
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if (enable && send_codec_inst_ &&
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codec_stack_params_.red_payload_types.count(send_codec_inst_->plfreq) <
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2015-11-23 04:30:52 -08:00
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1) {
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2015-05-07 15:49:23 +02:00
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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2015-12-16 04:19:08 -08:00
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"Cannot enable RED at %i Hz.", send_codec_inst_->plfreq);
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2015-03-30 19:00:44 +02:00
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return false;
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}
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2015-12-16 04:19:08 -08:00
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codec_stack_params_.use_red = enable;
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2015-03-30 19:00:44 +02:00
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return true;
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}
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2015-12-16 04:19:08 -08:00
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bool CodecManager::SetVAD(bool enable, ACMVADMode mode) {
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2015-03-30 19:00:44 +02:00
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// Sanity check of the mode.
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2015-09-17 00:24:34 -07:00
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RTC_DCHECK(mode == VADNormal || mode == VADLowBitrate || mode == VADAggr ||
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mode == VADVeryAggr);
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2015-03-30 19:00:44 +02:00
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// Check that the send codec is mono. We don't support VAD/DTX for stereo
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// sending.
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2015-12-15 14:21:33 -08:00
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const bool stereo_send =
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codec_stack_params_.speech_encoder
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? (codec_stack_params_.speech_encoder->NumChannels() != 1)
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: false;
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2015-11-10 09:47:36 -08:00
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if (enable && stereo_send) {
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"VAD/DTX not supported for stereo sending");
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2015-12-16 04:19:08 -08:00
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return false;
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2015-03-30 19:00:44 +02:00
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}
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2015-12-16 06:24:05 -08:00
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// TODO(kwiberg): This doesn't protect Opus when injected as an external
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// encoder.
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if (send_codec_inst_ && IsOpus(*send_codec_inst_)) {
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2015-12-16 04:19:08 -08:00
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// VAD/DTX not supported, but don't fail.
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enable = false;
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2015-05-07 15:49:23 +02:00
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}
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2015-12-16 04:19:08 -08:00
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codec_stack_params_.use_cng = enable;
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codec_stack_params_.vad_mode = mode;
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return true;
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2015-03-30 19:00:44 +02:00
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}
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2015-12-16 04:19:08 -08:00
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bool CodecManager::SetCodecFEC(bool enable_codec_fec) {
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2015-11-23 04:30:52 -08:00
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if (enable_codec_fec && codec_stack_params_.use_red) {
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2015-03-30 19:00:44 +02:00
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WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, 0,
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"Codec internal FEC and RED cannot be co-enabled.");
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2015-12-16 04:19:08 -08:00
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return false;
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2015-03-30 19:00:44 +02:00
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}
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2015-12-16 04:19:08 -08:00
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codec_stack_params_.use_codec_fec = enable_codec_fec;
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return true;
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2015-03-30 19:00:44 +02:00
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}
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2016-06-23 03:58:36 -07:00
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bool CodecManager::MakeEncoder(RentACodec* rac, AudioCodingModule* acm) {
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RTC_DCHECK(rac);
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RTC_DCHECK(acm);
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if (!recreate_encoder_) {
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bool error = false;
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|
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// Try to re-use the speech encoder we've given to the ACM.
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|
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acm->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
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|
|
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if (!*encoder) {
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|
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// There is no existing encoder.
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|
|
|
|
recreate_encoder_ = true;
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|
|
|
|
return;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
// Extract the speech encoder from the ACM.
|
|
|
|
|
std::unique_ptr<AudioEncoder> enc = std::move(*encoder);
|
|
|
|
|
while (true) {
|
|
|
|
|
auto sub_enc = enc->ReclaimContainedEncoders();
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|
|
|
|
if (sub_enc.empty()) {
|
|
|
|
|
break;
|
|
|
|
|
}
|
|
|
|
|
RTC_CHECK_EQ(1u, sub_enc.size());
|
|
|
|
|
|
|
|
|
|
// Replace enc with its sub encoder. We need to put the sub encoder in
|
|
|
|
|
// a temporary first, since otherwise the old value of enc would be
|
|
|
|
|
// destroyed before the new value got assigned, which would be bad
|
|
|
|
|
// since the new value is a part of the old value.
|
|
|
|
|
auto tmp_enc = std::move(sub_enc[0]);
|
|
|
|
|
enc = std::move(tmp_enc);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
// Wrap it in a new encoder stack and put it back.
|
|
|
|
|
codec_stack_params_.speech_encoder = std::move(enc);
|
|
|
|
|
*encoder = rac->RentEncoderStack(&codec_stack_params_);
|
|
|
|
|
if (!*encoder) {
|
|
|
|
|
error = true;
|
|
|
|
|
}
|
|
|
|
|
});
|
|
|
|
|
if (error) {
|
|
|
|
|
return false;
|
|
|
|
|
}
|
|
|
|
|
if (!recreate_encoder_) {
|
|
|
|
|
return true;
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
if (!send_codec_inst_) {
|
|
|
|
|
// We don't have the information we need to create a new speech encoder.
|
|
|
|
|
// (This is not an error.)
|
|
|
|
|
return true;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
codec_stack_params_.speech_encoder = rac->RentEncoder(*send_codec_inst_);
|
|
|
|
|
auto stack = rac->RentEncoderStack(&codec_stack_params_);
|
|
|
|
|
if (!stack) {
|
|
|
|
|
return false;
|
|
|
|
|
}
|
|
|
|
|
acm->SetEncoder(std::move(stack));
|
|
|
|
|
recreate_encoder_ = false;
|
|
|
|
|
return true;
|
|
|
|
|
}
|
|
|
|
|
|
2015-03-30 19:00:44 +02:00
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|
|
} // namespace acm2
|
|
|
|
|
} // namespace webrtc
|